<div>A example dialplan is shown below:</div>
<div> </div>
<div>exten => 5003,1,Answer()<br>exten => 5003,n,Rtsp(rtsp://<a href="http://192.168.1.1/live.sdp">192.168.1.1/live.sdp</a>)<br>exten => 5003,n,Hangup()<br><br>That means some caller(such as 5001) call 5003 will follow these steps?</div>
<div>1,answer ()</div>
<div>2,receive rtsp stream from rtsp://<a href="http://192.168.1.1/live.sdp">192.168.1.1/live.sdp</a></div>
<div>3,convert and play the stream to the caller(5001)</div>
<div>4,hangup ()</div>
<div> </div>
<div>Is that correct?<br></div>
<div class="gmail_quote">在 2010年6月24日 下午9:23，Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>></span>写道：<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br><br>Am 24.06.2010 14:59, schrieb 李波:<br>
<div class="im">> Hi klaus,<br>><br>> Only receive rtsp?<br>> What does asterisk do when it use app_rtsp?<br>> Receive rtsp stream?<br><br></div>
<div class="im">Yes, the RTSP stream is received, and then by app_rtsp converted into<br>the Asterisk internal format.<br><br></div>
<div class="im">> Not play the stream to the client?<br><br></div>
<div>
<div></div>
<div class="h5">Once the audio is "inside" Asterisk you can do whatever Asterisk can do<br>with Audio, e.g. send it to a SIP client as RTP stream (but you can not<br>send RTSP - but I think you do not even want this).<br>
<br>PS: Please stay on the mailing list<br><br>regards<br>Klaus<br><br></div></div></blockquote></div><br>